Re: [asterisk-users] multiple PRI's in one group ..how??
Thanks a lot.. ALl the information i need and much more:) very useful:) Oguzhan Kayhan escribió: I mean..making a single trunk between a pstn or telco with 2 or more PRI's.. I mean instead of using 32 channels to use 64 or more.. I am trying to increase the capacity between my PSTN and asterisk actually. There will be more than 35-40 concurrent calls so while creating a zap trunk(or dahdi..whatever u call) i want all pris to behave like they are a single span I hope i make myself clear now :) sorry for misunderstanding. Just put all channels of all spans you want on the same group (look at this example snip of zapata.conf): group=1 signalling = pri_cpe context=4pri channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 That way asterisk will use the 120 channels as one big trunk on g1. if you Dial(Zap/g1/number), it will use all the channels. If you want to do round-robin so it goes around all channels and not the first free, try Dial(Zap/r1/number) from incrementing round-robin, or Dial(Zap/R1/number) to decrementing round-robin. Of course, depending on your need you can split them off on the arrange you want, not only at a entire PRI level. If you wanted to reserve 5 channels to inbound calls only and the rest for outbound, you could do this for example: group=1 signalling = pri_cpe context=outbound-group channel = 6-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 group=7 signalling = pri_cpe context=inbound-group channel = 1-5 This way, if you Dial group 1, you won't use channels 1-5, leaving them free for inbound calls. Cheers, Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Opinion on Attended transfer in features.conf
Hi, In 1.6.1, it seems Attended Transfer do not behave exactly behave like Blind Transfer when transferer hangs up before callee answers : - in Blind Transfer, caller (ie transferee) is hearing Ringing tone when callee's phone is ringing - in Attended Transfer, caller (ie transferee) is hearing Music On Hold when callee's phone is ringing - in Attended Transfer, if callee don't answer in a given time frame, call comes back to transferer again Bottom line is you still need to teach both Blind and Attended transfers. Is there a way to set Attended Transfer to mimic exactly Blind Transfer ? What could be a use case in which one would need Attended Transfer not to transform into a Blind Transfer ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] setting codecs on the fly
Hi I would like the option to set the codec used on a call by call basis. I have a tdm410 2fxs + 1fxo. when I make calls to my vsp, they go through as ulaw, I am guessing because I have allowed if for the vsp (g729, alaw and ulaw). I would prefer to use g729 from the fxs to the vsp but I would like the option to make alaw calls on demand and maybe ulaw. Or even to set the default on the digium card to alaw - that seems to be the default in OZ Thanks signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to remove a GLOBAL variable from diaplan ?
Hello, Is there a way to remove a global variable from dialplan ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Function IMPORT
Hi, I've just discovered IMPORT function existence. It can be use to import values from channel's Variable section but unfortunately, il can't be use to access to values from Info section (I'm referring here to sections Info and Variables dumped by DumpChan application). Is there a way to work around this and access from one channel for instance to another channel's CallerIDNum variable ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - Aastra - mapping transfer key
Hi, I've read this : http://www.trixbox.org/forums/vendor-forums-certified/aastra-endpoints/9143i-re-map-transfer-key Has anyone tried this ? I would be very happy to get few more details on how to do this. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] external RTP IP address
Hello, I have asterisk 1.6.1.1 box behind NAT. On the same local network I've SIP proxy server too. The problem appears with RTP.My provider's RTP IP addresses are public. When asterisk sends SIP invite to SIP proxy, it defines local RTP IP, but not externIP. Maybe somebody knows how to solve this problem? Thanks -- Pagarbiai / Best Regards, Giedrius ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Queue Problem
You could try this one: http://www.voip-info.org/wiki/view/Asterisk+func+Devstate If I can add a warning, be wary of having both ACD (Queue) and non-ACD traffic on the same operator - you risk having awful performance. Just my two eurocents, l. 2009/6/12 Lee, John (Sydney) john@compuware.com I am running Asterisk 1.4.21.2 For reception, I defined a simple queue with one SIP phone as the only member. When I receive an incoming call, I test QUEUE_WAITING_COUNT to see if it is 0. If it is 0, then I will playback a message to tell the caller to be patient and then do a Queue(queue-name). If QUEUE_WAITING_COUNT is zero, then I will just Queue(queue-name, r) to ring the receptionist phone without playing any message. A problem arises if the receptionist is talking to someone on the phone. In this scenario, QUEUE_WAITING_COUNT is also zero but I will need to playback a pls-be-patient message as well. So, I need to find out whether the receptionist phone is busy even if QUEUE_WAITING_COUNT = 0. Do you know if there is anyway, without dialling a SIP channel, I can check if a SIP extension is engaged or not? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Open Source Soft Phone
Hi Guys, Any suggestions on any open source soft phones that has IAX and SIP support. I would also like to some programming over it and interface it with address book or LDAP in order to make the call making easier for the users. Thanks Manoj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
On 15 Jun 2009, at 12:05, Manoj Panicker - FOES wrote: Any suggestions on any open source soft phones that has IAX and SIP support. I would also like to some programming over it and interface it with address book or LDAP in order to make the call making easier for the users. Don't double post. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Click-to-dial CTI for Windows
Hello guys, Is there a decent click-to-dial CTI which works well with Asterisk? We have vanilla asterisk implementation and I have tried a few (ADA, Outcall etc) but they have poor documentation and don't work very well. We are looking for an application which can allow us to dial a number from Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls with call history using a hardware deskphone. It seems simple - but nothing so far fits the bill. Can you recommend something? Thanks! Milen The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. Compuware Limited (company number 1522537) is a company registered in England and Wales whose registered office is at 163 Bath Road, Slough SL1 4AA, Berkshire, United Kingdom. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
twinkle. 2009/6/15 Manoj Panicker - FOES manoj.panic...@emirates.com Hi Guys, Any suggestions on any open source soft phones that has IAX and SIP support. I would also like to some programming over it and interface it with address book or LDAP in order to make the call making easier for the users. Thanks Manoj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
I'm currently using Ekiga. I don't think I'd reccomend it though; it lacks a lot of basic features. -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
On Mon, Jun 15, 2009 at 12:51:25PM +0100, Geraint Lee wrote: twinkle. Twingle is a good SIP phone. But does not support IAX. At the moment the only one I can think of is yate-gtk :-) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click-to-dial CTI for Windows
Hi, I try Noojee Click and Outcall, and for my context they work fine. Some times ago I tried SanpANumber, but it was bought by Digium and substitute with ADA. Bye Marco 2009/6/15 Stefanov, Milen milen.stefa...@compuware.com Hello guys, Is there a decent click-to-dial CTI which works well with Asterisk? We have vanilla asterisk implementation and I have tried a few (ADA, Outcall etc) but they have poor documentation and don’t work very well. We are looking for an application which can allow us to dial a number from Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls with call history using a hardware deskphone. It seems simple - but nothing so far fits the bill. Can you recommend something? Thanks! Milen The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. Compuware Limited (company number 1522537) is a company registered in England and Wales whose registered office is at 163 Bath Road, Slough SL1 4AA, Berkshire, United Kingdom. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suggest Multi-tenant Hosted PBX ?
Hello All, We have a requirement of hosted multi-tenant PBX where we can map DID for different clients. Each client should have saperate interface of Reporting, Call Recordings, Voice Mail and other features. Please suggest some solution or let us know if have it to sell ? Regards, Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kas...@haditelecom.com MSN: kashif__na...@hotmail.com Gmail: meet.kas...@gmail.com Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
Excuse me? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 15 June 2009 15:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Open Source Soft Phone On 15 Jun 2009, at 12:05, Manoj Panicker - FOES wrote: Any suggestions on any open source soft phones that has IAX and SIP support. I would also like to some programming over it and interface it with address book or LDAP in order to make the call making easier for the users. Don't double post. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
On 15 Jun 2009, at 13:36, Manoj Panicker - FOES wrote: Excuse me? You sent this message twice. Send it once, and wait for a reply. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PrivacyManager no longer working properly
Quoting Jaap Winius jwin...@umrk.to: Previously, I had the PrivacyManager working for me exactly as would be expected, but after upgrading the OS to Debian lenny and Asterisk to v1.4.21.2 that's no longer the case. Anonymous callers are still confronted with the PrivacyManager, but now no matter what I set the minlength value to, e.g.: exten = jaap,n,PrivacyManager(1,1) ... (I'm not using a privacy.conf file), the submitted caller ID is always considered invalid. This issue has been resolved, at least on my system. After running some more tests, I discovered that the PrivacyManager was only having problems with calls coming in via SIP; anonymous calls incoming via ISDN were treated normally. The Asterisk version I was using was from Xorcom (1.4.21.2~dfsg-3 for Debian lenny). Thinking that the version might be a problem, I first decided to try for an upgrade. I noticed that Xorcom had a major update in store for me -- Asterisk v1.6.1.0~dfsg-1 -- but worried that the corresponding replacement of zaptel with dahdi software would cause problems (I need it to support my HFC-PCI card). Nevertheless, I gave it a try. Bad idea. I wasted several hours late last night trying to get the HFC-PCI card working working with dahdi, but without any luck. The first thing I noticed was that the zaphfc module was still there (not renamed), while the one that I prefer -- vzaphfc -- was not. To get dahdi_genconf to work I found that it was important for dahdi_dummy be loaded after zaphfc. That went fine, but then running dahdi_genconf would lock up the system, with thousands of error messages flashing across the server console: zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled. After a few of these lock-ups and reboots, I abandoned the upgrade. Obviously, I'll try for it again at a later date, but I really do hope that by that time I will discover that dahdi includes a working equivalent of vzaphfc. Not wanting to go against the grain by attempting to manually reinstall and then freezing the older asterisk and zaptel packages from Xorcom, which would certainly get me nowhere as far as my privacymanager problem was concerned, I decided at this point to try to install the stock version that comes with Debian lenny instead. After installing all of the necessary packages, I saw that the HFC-PCI card was working again, but so was the privacymanager (for both ISDN and SIP). All of my problems were solved! In hindsight, however, I see that I've been running the stock Debian versions of Asterisk and Zaptel for lenny all along. I was running v1.4.21.2~dfsg-3 before, just as I am now, but since Xorcom was until recently only offering an older version for Debian lenny, 1.4.21.1~dfsg-0.5941, apt wasn't selecting it. The same can be said for the Zaptel packages that I have installed now compared to before (1.4.11~dfsg-3), except that before I also had an even older zaptel-firmware package installed, 1.4.10.1-0.567, which must have come from Xorcom. I don't think that it was influencing matters, though, since the compiled zaptel-modules packages are still the same version now as before. So, how come the privacymanager is working 100% now? No idea. Thanks to my fantastic backup system, I'm also using the same Asterisk configuration files now as I was before. It's a mystery I guess. In the mean time, I will see if I can acquire an extra HFC-PCI card from somewhere and set up a new system with which to test Asterisk 1.6. Cheers, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Queue Problem
I posted a simple PERL agi that uses hints to do a similar thing to Devstate
last week. Here it is:
#!/usr/bin/perl
use strict;
use warnings;
# define variables
# show hints will get hint information from the dialplan
my $cmda = '/usr/sbin/asterisk -rx show hints ';
my $towatch = $ARGV[0];
# turn off I/O buffering
$| = 1;
# read the AGI environment
while (STDIN) {
chomp($_);
last if 0 == length($_);
}
# assume idle
print STDOUT SET VARIABLE LINESTAT \Idle\\n;
STDIN;
# get trunk information
$SIG{'PIPE'} = 'IGNORE';
open (my $trunk_info, $cmda) or exit;
while ($trunk_info) {
if (($_ =~ /internal/) ($_ =~ /$towatch/)) {
my $c = unpack(x74 a16, $_);
$c =~ s/\s//gx;
print STDOUT SET VARIABLE LINESTAT \$c\\n;
STDIN;
}
}
close $trunk_info;
Dialplan: exten = 2100,1,Noop(dial 102 after checking sippeer)
exten = 2100,n,Set(LINESTAT=Idle)
exten = 2100,n,AGI(steve.agi|102)
exten = 2100,n,Wait(3)
exten = 2100,n,Verbose(status is ${LINESTAT})
exten = 2100,n,Gotoif($[${LINESTAT} != Idle]?inuse)
exten = 2100,n,Dial(SIP/102,20,m)
exten = 2100,n,Background(vm-goodbye)
exten = 2100,n,Hangup
exten = 2100,n(inuse),Voicemail(1...@default)
exten = 2100,n,Background(vm-goodbye)
just change 102 to your receptionists number
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Monday, June 15, 2009 5:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Simple Queue Problem
You could try this one:
http://www.voip-info.org/wiki/view/Asterisk+func+Devstate
If I can add a warning, be wary of having both ACD (Queue) and non-ACD
traffic on the same operator - you risk having awful performance.
Just my two eurocents,
l.
2009/6/12 Lee, John (Sydney) john@compuware.com
I am running Asterisk 1.4.21.2
For reception, I defined a simple queue with one SIP phone as the only
member.
When I receive an incoming call, I test QUEUE_WAITING_COUNT to see if it
is 0.
If it is 0, then I will playback a message to tell the caller to be
patient and then do a Queue(queue-name).
If QUEUE_WAITING_COUNT is zero, then I will just Queue(queue-name, r)
to ring the receptionist phone without playing any message.
A problem arises if the receptionist is talking to someone on the phone.
In this scenario, QUEUE_WAITING_COUNT is also zero but I will need to
playback a pls-be-patient message as well.
So, I need to find out whether the receptionist phone is busy even if
QUEUE_WAITING_COUNT = 0.
Do you know if there is anyway, without dialling a SIP channel, I can
check if a SIP extension is engaged or not?
___
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--
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] How to remove a GLOBAL variable from diaplan ?
Remove the Set in extensions.conf and reload the dialplan. If you don't have that capability, just do a set to null. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Monday, June 15, 2009 4:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to remove a GLOBAL variable from diaplan ? Hello, Is there a way to remove a global variable from dialplan ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinion on Attended transfer in features.conf
Olivier wrote: Hi, In 1.6.1, it seems Attended Transfer do not behave exactly behave like Blind Transfer when transferer hangs up before callee answers : - in Blind Transfer, caller (ie transferee) is hearing Ringing tone when callee's phone is ringing - in Attended Transfer, caller (ie transferee) is hearing Music On Hold when callee's phone is ringing - in Attended Transfer, if callee don't answer in a given time frame, call comes back to transferer again Bottom line is you still need to teach both Blind and Attended transfers. Is there a way to set Attended Transfer to mimic exactly Blind Transfer ? What could be a use case in which one would need Attended Transfer not to transform into a Blind Transfer ? I have wondered for years now why someone thought there needed to be two different transfer functions. Transfer should be ONE function. If one wants to speak first to the object of the transfer, then stay until they answer, otherwise hang up and the transfer is completed. Two independent transfers that have to start with different codes is just awkward and dumb and long ago needed to be fixed. I suppose it started life because someone had a weak knowledge of basic telephony, but I really don't know. Learn from history and improve on it. When one reinvents the wheel, sometimes one ends up with an ellipse. JMO John Novack Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and google talk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I try to set up a gateway gtalk to sip.
I test asterisk 1.6.1 and 1.4.21 from debian repository and the result
is identical : no sound during the call.
my jabber.conf :
[general]
debug=yes
autoprune=no
autoregister=no
[allo-gw]
type=client
serverhost=talk.google.com
username=xx.xa...@gmail.com/asterisk
secret=Xxxx
port=5222
usetls=yes
usesasl=yes
statusmessage=I am available
timeout=100
my gtalk.conf :
[general]
context=gtalk
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=gtalk
my extensions.conf :
[gtalk]
exten = s,1,Answer()
exten = s,n,Dial(Local/1...@internal)
exten = s,n,Hangup()
[internal]
exten = 100X,1,Answer()
exten = 100X,n,Dial(${EXTEN})
exten = 100X,n,Hangup()
thanks for your idea
- --
Antoine Patte
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Version: GnuPG v1.4.9 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iEYEARECAAYFAko2UDsACgkQBnIOcv+j7+yy4ACfWPaYjM2D/sFJZr6l/6l0NimY
fVQAnjvv/22KLQmFvHmY16SUGAfW96OU
=9VvC
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Re: [asterisk-users] Click-to-dial CTI for Windows
I use snapanumber for dialing from Outlook works great. Don't know what Digium did to it when they made it Outcallbut you're not the only one who has said they had a problem with it. Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefanov, Milen Sent: Monday, June 15, 2009 7:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Click-to-dial CTI for Windows Hello guys, Is there a decent click-to-dial CTI which works well with Asterisk? We have vanilla asterisk implementation and I have tried a few (ADA, Outcall etc) but they have poor documentation and don't work very well. We are looking for an application which can allow us to dial a number from Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls with call history using a hardware deskphone. It seems simple - but nothing so far fits the bill. Can you recommend something? Thanks! Milen The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. Compuware Limited (company number 1522537) is a company registered in England and Wales whose registered office is at 163 Bath Road, Slough SL1 4AA, Berkshire, United Kingdom. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suggest Multi-tenant Predictive Dialer ?
Hello All, We have a requirement of multi-tenant Predictive Dialer which we can sell to multiple call centers. Each call center will have saperate interface for setting up campaigns and Reporting. Please suggest some solution or let us know if have it to sell ? Regards, Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kas...@haditelecom.com MSN: kashif__na...@hotmail.com Gmail: meet.kas...@gmail.com Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
On Monday, June 15, 2009, Steve Howes wrote: On 15 Jun 2009, at 13:36, Manoj Panicker - FOES wrote: Excuse me? You sent this message twice. Send it once, and wait for a reply. Only received once here. My mail server is configured to remove duplicated messages - but a different timestamp would make the two copies non-duplicated. IOW, it looks to me like the list server had a hiccough and Christopher wrongly accused the OP. -- Geoff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
On 15 Jun 2009, at 15:54, Geoff Lane wrote: On Monday, June 15, 2009, Steve Howes wrote: On 15 Jun 2009, at 13:36, Manoj Panicker - FOES wrote: Excuse me? You sent this message twice. Send it once, and wait for a reply. Only received once here. My mail server is configured to remove duplicated messages - but a different timestamp would make the two copies non-duplicated. IOW, it looks to me like the list server had a hiccough and Christopher wrongly accused the OP. AC5F42F85475254AAB74B5A2B0653E440212BBAF @DXBHQMBEX10.corp.emirates.com at 12:09 AC5F42F85475254AAB74B5A2B0653E440212B8F4 @DXBHQMBEX10.corp.emirates.com at 19:16 yesterday Both different submission times. Definitely sent as two messages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Open Source Call Statistics / Metrics Packages
Hi, Just wondering what the popular open source call statistics / metrics packages are for Asterisk? Preferably an all-in-one package that supports queues and calls from the CDR information generated by Asterisk. Whats everyone using? Favorites? Thanks, Marc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
On Mon, Jun 15, 2009 at 10:54 AM, Geoff Lane ge...@gjctech.co.uk wrote: Only received once here. Only once here also, using gmail. IOW, it looks to me like the list server had a hiccough and Christopher wrongly accused the OP. Steve did the 'accusing', not me... ;-) -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sending sip info messages
Thanks for your information! Now I tried to send a Sip Messages, instead of a Sip Info. Between two softphones the exchange of sip messages works fine. But the message relay over the asterisk doesn't work: Status 415 Unsupported Media Type Does someone know, how to activate the exchange of sip messages inside asterisk? Perhaps I have to change the sip.conf file or is the relay not possible? Many thanks in advance! Regards Karsten -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug or feature : how to customize SIP REFER from dialplan
Hi, I've been editing my dialplan to launch custom instructions anytime a SIP REFER-based transfer occurs. The only hook I could find is catching an hangup event which is tied to a Zombie channel (ie a channel named like SIP/1234-vhvebjvnvZOMBIE). Is this a feature or a bug ? In other words, do you think : - it shouldn't be possible at all to hook custom instructions for SIP REFER-based transfer occurs (then I obviously found a bug), - catching ZOMBIE channel hangup is the way to hook custom instructions for a SIP REFER-based transfer. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to remove a GLOBAL variable from diaplan ?
On Monday 15 June 2009 04:06:31 am Olivier wrote: Is there a way to remove a global variable from dialplan ? Set(GLOBAL(foo)=) -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Function IMPORT
On Monday 15 June 2009 04:03:48 am Olivier wrote:
I've just discovered IMPORT function existence.
It can be use to import values from channel's Variable section but
unfortunately, il can't be use to access to values from Info section
(I'm referring here to sections Info and Variables dumped by DumpChan
application).
You're mistaken.
Is there a way to work around this and access from one channel for instance
to another channel's CallerIDNum variable ?
${IMPORT(SIP/foo-abcd1234,CALLERID(num))} works fine.
--
Tilghman
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[asterisk-users] Asterisk 1.6.2.0-beta3 Now Available
The Asterisk Development Team is pleased to announce the third beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta3 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This is an incremental release of the 1.6.2.0 branch as the previous beta was released just over a month ago, and many issues have been resolved since then. Included in this release are the following issues reported by the community: * Update spiral support in trunk and 1.6.x branches to match what is in 1.4 (related to issue #13630). * Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over (issue #14815). * Fix a bug where the codecs of the called party leg were not properly sent back to the call leg when reinvited (issue #13569). * Fix broken attended transfers (issue #15183). * Add flags to chanspy audiohook so that audio stays in sync (issue #13745). * Resolve issues with choppy sound when using res_timing_pthread (issue #14412) Additionally, an update to chan_iax2 related to issue AST-2009-001 is included in this beta release. For more information, see: http://downloads.asterisk.org/pub/security/AST-2009-001.html For a full list of changes in this beta, please see the ChangeLog: http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/ChangeLog You can get more information about the new features and various changes in Asterisk 1.6.2.0 at: http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/CHANGES And if you're upgrading from previous versions of Asterisk see this file: http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/UPGRADE.txt Issues discovered in testing of this beta can be reported at http://issues.asterisk.org Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie, Question on making a PSTN call..
Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. On my Ubuntu: Next up, I connected my phone(NOKIA N73) to my computer, ensured that I am able to connect to internet on my ubuntu. wvdial works good too. Again, I am unsure how to get asterisk to connect to this modem so that I can use my soft phones to make a call. Need help. Thanks in Advance. -- Shivku, http://blog.shivku.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggest Multi-tenant Predictive Dialer ?
Kashif This changes things. We can do, but this is hardly a simple off-the- shelf. I will call you midday Tuesday if I might Steve On Jun 15, 2009, at 3:46 PM, Kashif Naeem wrote: Hello All, We have a requirement of multi-tenant Predictive Dialer which we can sell to multiple call centers. Each call center will have saperate interface for setting up campaigns and Reporting. Please suggest some solution or let us know if have it to sell ? Regards, Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kas...@haditelecom.com MSN: kashif__na...@hotmail.com Gmail: meet.kas...@gmail.com Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom registration errors
On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote: [hft0] type=friend username=hft0 secret=mysecret context=outtrunk-office host=192.168.200.99 Change the above to host=dynamic I just did this and did a 'reload'. reg.1.server.1.address=jtsd05 Can the phone resolve this unqualified name? Yes. It's in the search path, but just to be sure I put in an FQDN. Still, no change :-( ... chan_sip.c: Registration from 'sip:6193644...@jtsd05.nccom.com' failed for '192.168.200.99' - Username/auth name mismatch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
Well, lets just take the OP out and shoot him! GEESH! Can we all just move on, or MUST we waste more and more time and messages sent to reportedly 10,000 people on this unimportant issue. The original responder could have simply answered the guy's question or even better said nothing, instead of acting as self appointed list police, which we certainly don't need even more of! Peg Leg O'Brien Christopher Stamper wrote: On Mon, Jun 15, 2009 at 10:54 AM, Geoff Lane ge...@gjctech.co.uk mailto:ge...@gjctech.co.uk wrote: Only received once here. Only once here also, using gmail. IOW, it looks to me like the list server had a hiccough and Christopher wrongly accused the OP. Steve did the 'accusing', not me... ;-) -- Christopher Stamper Email: christopherstam...@gmail.com mailto:christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom registration errors
On Mon, 15 Jun 2009, Jim Gottlieb wrote: On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote: [hft0] type=friend username=hft0 secret=mysecret context=outtrunk-office host=192.168.200.99 Change the above to host=dynamic I just did this and did a 'reload'. reg.1.server.1.address=jtsd05 Can the phone resolve this unqualified name? Yes. It's in the search path, but just to be sure I put in an FQDN. Still, no change :-( ... chan_sip.c: Registration from 'sip:6193644...@jtsd05.nccom.com' failed for '192.168.200.99' - Username/auth name mismatch I am a bit confused as to the names and addresses involved here. Which name/address is the server, and which is the phone? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom registration errors
Pardon my ignorance, but can you register the external sip name to your internal ip (192.168.x.x)? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Monday, June 15, 2009 2:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom registration errors On Mon, 15 Jun 2009, Jim Gottlieb wrote: On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote: [hft0] type=friend username=hft0 secret=mysecret context=outtrunk-office host=192.168.200.99 Change the above to host=dynamic I just did this and did a 'reload'. reg.1.server.1.address=jtsd05 Can the phone resolve this unqualified name? Yes. It's in the search path, but just to be sure I put in an FQDN. Still, no change :-( ... chan_sip.c: Registration from 'sip:6193644...@jtsd05.nccom.com' failed for '192.168.200.99' - Username/auth name mismatch I am a bit confused as to the names and addresses involved here. Which name/address is the server, and which is the phone? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom registration errors
On 2009-06-15 at 19:12, Jeff LaCoursiere (j...@jeff.net) wrote: chan_sip.c: Registration from 'sip:6193644...@jtsd05.nccom.com' failed for '192.168.200.99' - Username/auth name mismatch I am a bit confused as to the names and addresses involved here. Which name/address is the server, and which is the phone? The phone is 192.168.200.99. The server is jtsd05. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom registration errors
Jim Gottlieb wrote: I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from 'sip:6193644...@jtsd05' failed for '192.168.200.99' - Username/auth name mismatch Turning on SIP debug, it appears it's asterisk trying to register with the phone: Using latest REGISTER request as basis request Sending to 192.168.200.99 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.200.99:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.200.99;branch=z9hG4bKd732a3486F528693;received=192.168.200.99 From: 6193644850 sip:6193644...@jtsd05;tag=A1BB38FF-7161AAEA To: sip:6193644...@jtsd05;tag=as3d68239c Call-ID: 20f907fe-db323389-f4569...@192.168.200.99 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 But then, the From: and To: lines seem to both show it from hostname jtsd05, though there's also the line saying it's going to 192.168.200.99 (the phone). I've played with all sorts of settings in sip.conf, but the messages persist. Here's what I've got: [hft0] type=friend username=hft0 secret=mysecret context=outtrunk-office host=192.168.200.99 disallow=all allow=ulaw dtmfmode=rfc2833 progressinband=no ;Polycom phones have trouble with the progressinband=never callerid=HFT Booth 0 (619) 364-4850 allowsubscribe=yes And some of the Polycom phone config: reg reg.1.displayName=HFT0 reg.1.address=6193644850 reg.1.label=4850 reg.1.type=private reg.1.lcs= reg.1.csta= reg.1.thirdPartyName= reg.1.auth.userId=hft0 reg.1.auth.password=mysecret reg.1.auth.optimizedInFailover= reg.1.musicOnHold.uri= reg.1.server.1.address=jtsd05 reg.1.server.1.port= reg.1.server.1.transport=DNSnaptr reg.1.server.2.transport=DNSnaptr reg.1.server.1.expires= reg.1.server.1.expires.overlap= reg.1.server.1.register= reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount= reg.1.server.1.expires.lineSeize= reg.1.server.1.lcs= reg.1.outboundProxy.address= Try changing reg.1.address to hft0. My hunch is asterisk is looking at the from of 6193644...@jtsd05 and going huh? I don't know a 6193644...@jtsd05. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom registration errors
On 2009-06-15 at 17:04, Dave Fullerton (dfullertaster...@shorelinecontainer.com) wrote: Try changing reg.1.address to hft0. My hunch is asterisk is looking at the from of 6193644...@jtsd05 and going huh? I don't know a 6193644...@jtsd05. That makes sense and it fixed it. Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tdm loosing interrupts and latency
Hi I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). I am using the box as a firewall/asterisk box. The problem occurs when I drop ppp and I get dead loop dectiotn going, I seem to lose interrupts and get lots of messages in syslog from wctdm24xx saying missed interrupt increasing latency its out lined here (http://forums.digium.com/viewtopic.php?p=126997highlight=sid=9de59f41f1a93ee8701b28fdd0cf6073) Seems like the driver (and this is in zaptel dadhi code), increases latency by +1 until 30. and then the card seems to not work. In my case I have seen latency increase from 8m (I have this as a starting point in the module load) up to 17ms usually around here the fxs and fxo ports stop working . I have to unload and then reload the module. bummer. I can think of a couple of solutions 1) build some intelligence to bring down the number when things are okay 2) build logic to say if a number is provided on module load to fix it to that 3) add a sysfs (/proc) interface to allow changing this value on the fly I could also try and solve my problem with the dead loop detection cat /proc/interrupts CPU0 0: 23809265XT-PIC-XTtimer 1: 0XT-PIC-XTi8042 2: 0XT-PIC-XTcascade 4:255XT-PIC-XTserial 5: 459544XT-PIC-XTeth1 8: 0XT-PIC-XTrtc0 10: 95177163XT-PIC-XTwctdm24xxp0 11: 28938443XT-PIC-XTeth0 12: 28938632XT-PIC-XTeth3 14:3624228XT-PIC-XTide0 15: 1XT-PIC-XTehci_hcd:usb1, ohci_hcd:usb2 NMI: 0 Non-maskable interrupts LOC: 0 Local timer interrupts TRM: 0 Thermal event interrupts SPU: 0 Spurious interrupts ERR: 0 MIS: 0 as you can see with the interrupts the wctdm24xxp0 is above eth0 (local lan) and eth3 (my adsl) eth1 is wireless and not heavily used So any one had this problems, any other possible solution to this ? How to engage digium to providing a fix for this ? Alex signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm loosing interrupts and latency
Alex Samad wrote: Hi I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). I am using the box as a firewall/asterisk box. The problem occurs when I drop ppp and I get dead loop dectiotn going, I seem to lose interrupts and get lots of messages in syslog from wctdm24xx saying missed interrupt increasing latency its out lined here (http://forums.digium.com/viewtopic.php?p=126997highlight=sid=9de59f41f1a93ee8701b28fdd0cf6073) Seems like the driver (and this is in zaptel dadhi code), increases latency by +1 until 30. and then the card seems to not work. In my case I have seen latency increase from 8m (I have this as a starting point in the module load) up to 17ms usually around here the fxs and fxo ports stop working . I have to unload and then reload the module. bummer. I can think of a couple of solutions 1) build some intelligence to bring down the number when things are okay 2) build logic to say if a number is provided on module load to fix it to that 3) add a sysfs (/proc) interface to allow changing this value on the fly I could also try and solve my problem with the dead loop detection cat /proc/interrupts CPU0 0: 23809265XT-PIC-XTtimer 1: 0XT-PIC-XTi8042 2: 0XT-PIC-XTcascade 4:255XT-PIC-XTserial 5: 459544XT-PIC-XTeth1 8: 0XT-PIC-XTrtc0 10: 95177163XT-PIC-XTwctdm24xxp0 11: 28938443XT-PIC-XTeth0 12: 28938632XT-PIC-XTeth3 14:3624228XT-PIC-XTide0 15: 1XT-PIC-XTehci_hcd:usb1, ohci_hcd:usb2 NMI: 0 Non-maskable interrupts LOC: 0 Local timer interrupts TRM: 0 Thermal event interrupts SPU: 0 Spurious interrupts ERR: 0 MIS: 0 as you can see with the interrupts the wctdm24xxp0 is above eth0 (local lan) and eth3 (my adsl) eth1 is wireless and not heavily used So any one had this problems, any other possible solution to this ? How to engage digium to providing a fix for this ? Alex If your ppp is dropping, that means you have lost Internet connectivity, correct? If that is the case, then that is your problem as Asterisk does not tolerate the lose of DNS resolution very well. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging
I have a production PBX (1.6.0.9) 150+ phones with MS Exchange 2007 and OCS working very well out of the box. We're using SIP/TCP support in 1.6.x; Believe it or not the most challenging part is to get MWI signaling back from Exchange. Let me know if I can help. Jim j...@sigma-networks.com; 408-701-9929 - Original Message - From: Wayne wa...@planetwayne.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 11, 2009 10:10:05 AM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging David Backeberg wrote: I would ask the question the other way around. Are there any plans for Microsoft to release a unified messaging product that will comply with SIP over UDP? I do see your point in a potential (ok who are kidding - real) risk of a system crash with using MS having full control over your phone system but, I was thinking along the lines of using exchange really only as a messaging system - ie voice mail, email reader. From what I can make out MS are even going along the lines of doing speech to text with 2010 version (I think it has text to speech already). I would have to agree that the PBX side of things is held still by Asterisk and I don't see my view on that changing yet, but, I would imagine MS would dig their heels in rather than changing exchange. The Asterisk community, being more open minded to change, could easily(?) make this work. Thanks Wayne. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
All right Steve Thanks. I thought it never went. My apologies. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 15 June 2009 16:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Open Source Soft Phone On 15 Jun 2009, at 13:36, Manoj Panicker - FOES wrote: Excuse me? You sent this message twice. Send it once, and wait for a reply. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm loosing interrupts and latency
On Mon, Jun 15, 2009 at 08:19:33PM -0500, Lyle Giese wrote: Alex Samad wrote: Hi [snip] as you can see with the interrupts the wctdm24xxp0 is above eth0 (local lan) and eth3 (my adsl) eth1 is wireless and not heavily used So any one had this problems, any other possible solution to this ? How to engage digium to providing a fix for this ? Alex If your ppp is dropping, that means you have lost Internet connectivity, correct? If that is the case, then that is your problem as Asterisk does not tolerate the lose of DNS resolution very well. I think you have missed the point of the question. But I use and internal dns server, I understand if I lose my adsl my voip calls will be lost, but I also route some calls out pstn, they should stay. But the problem is with the digium driver not with asterisk (which make s me think this might not be the right mailing list !) Lyle Giese LCR Computer Services, Inc. -- The suicide bombings have increased. There's too many of them. - George W. Bush 08/15/2001 Albuquerque, NM signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No exten available after pass between servers
Hello List!
I have 2 asterisk servers, The Admin(.20), and the Call Center(.21).
The Admin server contains the 1XXX extension and the Call Center hosts
the 2XXX extensions. I would like for our Admin folks to be able to
call the Call Center folks (and vice versa).
The call will go over the server fine, but when the Call Center server
answer, the CLI returns:
NOTICE[4296]: chan_iax2.c:7398 socket_read: Rejected connect attempt
from 10.0.10.20, request '2...@2xxx' does not exist
sip show peers does register the phone:
2100/210010.0.10.237 D 5060 Unmonitored
-
Admin(.20)
exten = _2XXX,1,Answer()
exten = _2XXX,n,Dial(IAX2/2XXX/${EXTEN},20)
IAX.conf
[2XXX]
type=friend
username=2XXX
secret=
auth=plaintext
host=10.0.10.21
context=internal
qualify=yes
trunk=yes
-
Call Center(.21)
[2XXX]
exten = s,1,Answer()
exten = _2XXX,n,Dial(SIP/${EXTEN})
IAX.conf
[2XXX]
type=friend
username=2XXX
secret=
auth=plaintext
host=10.0.10.20
context=2XXX
qualify=yes
trunk=yes
-
The Call Center (.21) also hosts VICIDial if that would cause a
conflict with registration
Any input is much appreciated.
- Dan Pilcheck
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Re: [asterisk-users] No exten available after pass between servers
Dan Pilcheck wrote: The call will go over the server fine, but when the Call Center server answer, the CLI returns: NOTICE[4296]: chan_iax2.c:7398 socket_read: Rejected connect attempt from 10.0.10.20, request '2...@2xxx' does not exist What context are the phones in the extension range 2XXX in? I don't know what Vicidial's default context for extensions is, but I'd be surprised if it's 2XXX. Can you show us the results of sip show users after you remove the secret from the output? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm loosing interrupts and latency
On Monday 15 June 2009 20:00:11 Alex Samad wrote: I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). How to engage digium to providing a fix for this ? http://www.digium.com/en/supportcenter/ -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie, Question on making a PSTN call..
Need help pls..Anyone? On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote: Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. On my Ubuntu: Next up, I connected my phone(NOKIA N73) to my computer, ensured that I am able to connect to internet on my ubuntu. wvdial works good too. Again, I am unsure how to get asterisk to connect to this modem so that I can use my soft phones to make a call. Need help. Thanks in Advance. -- Shivku, http://blog.shivku.com -- Shivku, http://blog.shivku.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
